Add DeliveryStatus enum to DeliverPacket().
Allows signalling why packet delivery failed. Especially enables
signaling that delivery fails because the incoming packet had an unknown
SSRC. This allows an application to react and create receivers for the
new streams.
R=mflodman@webrtc.org
BUG=3228
Review URL: https://webrtc-codereview.appspot.com/12289005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6150 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/video/call.cc b/webrtc/video/call.cc
index a327654..6adb0fc 100644
--- a/webrtc/video/call.cc
+++ b/webrtc/video/call.cc
@@ -86,13 +86,14 @@
virtual uint32_t SendBitrateEstimate() OVERRIDE;
virtual uint32_t ReceiveBitrateEstimate() OVERRIDE;
- virtual bool DeliverPacket(const uint8_t* packet, size_t length) OVERRIDE;
+ virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
+ size_t length) OVERRIDE;
private:
- bool DeliverRtcp(const uint8_t* packet, size_t length);
- bool DeliverRtp(const RTPHeader& header,
- const uint8_t* packet,
- size_t length);
+ DeliveryStatus DeliverRtcp(const uint8_t* packet, size_t length);
+ DeliveryStatus DeliverRtp(const RTPHeader& header,
+ const uint8_t* packet,
+ size_t length);
Call::Config config_;
@@ -278,9 +279,12 @@
return 0;
}
-bool Call::DeliverRtcp(const uint8_t* packet, size_t length) {
+Call::PacketReceiver::DeliveryStatus Call::DeliverRtcp(const uint8_t* packet,
+ size_t length) {
// TODO(pbos): Figure out what channel needs it actually.
// Do NOT broadcast! Also make sure it's a valid packet.
+ // Return DELIVERY_UNKNOWN_SSRC if it can be determined that
+ // there's no receiver of the packet.
bool rtcp_delivered = false;
{
ReadLockScoped read_lock(*receive_lock_);
@@ -303,30 +307,33 @@
rtcp_delivered = true;
}
}
- return rtcp_delivered;
+ return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
}
-bool Call::DeliverRtp(const RTPHeader& header,
- const uint8_t* packet,
- size_t length) {
+Call::PacketReceiver::DeliveryStatus Call::DeliverRtp(const RTPHeader& header,
+ const uint8_t* packet,
+ size_t length) {
ReadLockScoped read_lock(*receive_lock_);
std::map<uint32_t, VideoReceiveStream*>::iterator it =
receive_ssrcs_.find(header.ssrc);
- if (it == receive_ssrcs_.end()) {
- // TODO(pbos): Log some warning, SSRC without receiver.
- return false;
- }
- return it->second->DeliverRtp(static_cast<const uint8_t*>(packet), length);
+
+ if (it == receive_ssrcs_.end())
+ return DELIVERY_UNKNOWN_SSRC;
+
+ return it->second->DeliverRtp(static_cast<const uint8_t*>(packet), length)
+ ? DELIVERY_OK
+ : DELIVERY_PACKET_ERROR;
}
-bool Call::DeliverPacket(const uint8_t* packet, size_t length) {
+Call::PacketReceiver::DeliveryStatus Call::DeliverPacket(const uint8_t* packet,
+ size_t length) {
// TODO(pbos): ExtensionMap if there are extensions.
if (RtpHeaderParser::IsRtcp(packet, static_cast<int>(length)))
return DeliverRtcp(packet, length);
RTPHeader rtp_header;
if (!rtp_header_parser_->Parse(packet, static_cast<int>(length), &rtp_header))
- return false;
+ return DELIVERY_PACKET_ERROR;
return DeliverRtp(rtp_header, packet, length);
}
diff --git a/webrtc/video/call_perf_tests.cc b/webrtc/video/call_perf_tests.cc
index 9830dd1..0076ce2 100644
--- a/webrtc/video/call_perf_tests.cc
+++ b/webrtc/video/call_perf_tests.cc
@@ -258,7 +258,8 @@
: channel_(channel),
voe_network_(voe_network),
parser_(RtpHeaderParser::Create()) {}
- virtual bool DeliverPacket(const uint8_t* packet, size_t length) {
+ virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
+ size_t length) OVERRIDE {
int ret;
if (parser_->IsRtcp(packet, static_cast<int>(length))) {
ret = voe_network_->ReceivedRTCPPacket(
@@ -267,7 +268,7 @@
ret = voe_network_->ReceivedRTPPacket(
channel_, packet, static_cast<unsigned int>(length), PacketTime());
}
- return ret == 0;
+ return ret == 0 ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
}
private:
@@ -589,7 +590,8 @@
}
private:
- virtual bool DeliverPacket(const uint8_t* packet, size_t length) OVERRIDE {
+ virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
+ size_t length) OVERRIDE {
VideoSendStream::Stats stats = send_stream_->GetStats();
if (stats.substreams.size() > 0) {
assert(stats.substreams.size() == 1);
diff --git a/webrtc/video/call_tests.cc b/webrtc/video/call_tests.cc
index 617b944..b305246 100644
--- a/webrtc/video/call_tests.cc
+++ b/webrtc/video/call_tests.cc
@@ -840,7 +840,7 @@
ReceivesPliAndRecovers(0);
}
-TEST_F(CallTest, SurvivesIncomingRtpPacketsToDestroyedReceiveStream) {
+TEST_F(CallTest, UnknownRtpPacketGivesUnknownSsrcReturnCode) {
class PacketInputObserver : public PacketReceiver {
public:
explicit PacketInputObserver(PacketReceiver* receiver)
@@ -851,13 +851,16 @@
}
private:
- virtual bool DeliverPacket(const uint8_t* packet, size_t length) {
+ virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
+ size_t length) OVERRIDE {
if (RtpHeaderParser::IsRtcp(packet, static_cast<int>(length))) {
return receiver_->DeliverPacket(packet, length);
} else {
- EXPECT_FALSE(receiver_->DeliverPacket(packet, length));
+ DeliveryStatus delivery_status =
+ receiver_->DeliverPacket(packet, length);
+ EXPECT_EQ(DELIVERY_UNKNOWN_SSRC, delivery_status);
delivered_packet_->Set();
- return false;
+ return delivery_status;
}
}
diff --git a/webrtc/video/full_stack.cc b/webrtc/video/full_stack.cc
index dbd234b..cb0ba55 100644
--- a/webrtc/video/full_stack.cc
+++ b/webrtc/video/full_stack.cc
@@ -111,7 +111,8 @@
virtual void SetReceiver(PacketReceiver* receiver) { receiver_ = receiver; }
- virtual bool DeliverPacket(const uint8_t* packet, size_t length) OVERRIDE {
+ virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
+ size_t length) OVERRIDE {
scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
RTPHeader header;
parser->Parse(packet, static_cast<int>(length), &header);
diff --git a/webrtc/video/rampup_tests.cc b/webrtc/video/rampup_tests.cc
index 32a2998..c86963d 100644
--- a/webrtc/video/rampup_tests.cc
+++ b/webrtc/video/rampup_tests.cc
@@ -268,7 +268,8 @@
return test::DirectTransport::SendRtp(data, length);
}
- virtual bool DeliverPacket(const uint8_t* packet, size_t length) OVERRIDE {
+ virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
+ size_t length) OVERRIDE {
CriticalSectionScoped lock(crit_.get());
RTPHeader header;
EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header));
@@ -279,7 +280,7 @@
remote_bitrate_estimator_->Process();
}
suspended_in_stats_ = send_stream_->GetStats().suspended;
- return true;
+ return DELIVERY_OK;
}
virtual bool SendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index 13ac01c..6662866 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -1160,13 +1160,14 @@
}
private:
- virtual bool DeliverPacket(const uint8_t* packet, size_t length) {
+ virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
+ size_t length) OVERRIDE {
if (RtpHeaderParser::IsRtcp(packet, static_cast<int>(length)))
- return true;
+ return DELIVERY_OK;
RTPHeader header;
if (!parser_->Parse(packet, static_cast<int>(length), &header))
- return true;
+ return DELIVERY_PACKET_ERROR;
assert(send_stream_ != NULL);
VideoSendStream::Stats stats = send_stream_->GetStats();
if (!stats.substreams.empty()) {
@@ -1188,7 +1189,7 @@
observation_complete_->Set();
}
}
- return true;
+ return DELIVERY_OK;
}
scoped_ptr<RtpRtcp> rtp_rtcp_;
